good.Internet programming is different because routers on the Internet may set the MTU to a different value. If we assume that the MTU is sending data to 1500来, and the MTU value of a network passing through is less than 1500 bytes, then the system will use a series of mechanisms to adjust the MTU value so that the datagram can reach its destination smoothly. Because the standard MTU value on the Internet is 576 bytes, it is best to use UDP data-length controls within 548 bytes (576-8-20) for U
UDP (packet length, packet collection capability, packet loss and process structure selection)UDP Packet Length: the theoretical length of a UDP packet
What is the theoretical length of udp data packets and what is the proper udp
between two routers, regardless of packet length or link transfer rate.
queue delay and packet loss
Queue Delay
ratio LA/R is called flow intensityIf la/r >1, the average rate at which the bit arrives in the queue exceeds the rate at which the queue is transferred, the increase in the queue tends to be unbounded, and the queuing delay tends to be infinite. The
in front of the Libpcap capture packet, especially in the gigabit network conditions, a large number of drops, online search for a long time, probably all Pf_packet +mmap,napi,pf_ring and other methods, I pf_ring+libpcap experiment, The detection of gigabit network conditions, capturing the performance of the packet is very good, almost no packet
Https://www.tuicool.com/articles/7ni2yyr
Recently work encountered a server application UDP lost packet, in the process of checking a lot of information, summed up this article for more people's reference.
Before we begin, we'll use a diagram to explain the process of receiving a network message from a Linux system. First of all, network packets sent through the physical network to the NIC driver will read the message in the network into the ring buf
UDP-based RTP transmission in the complex public network environment, especially 3G, 4G, WiFi network faced with packet loss, disorderly sequence, repetition, jitter and other issues, seriously affect the real-time audio and video interactive effect, even if it is a RTP packet lost, if the receiver does not do processing, will also lead to the appearance of video
Reprint: http://blog.csdn.net/galaxy_fxstar/article/details/5290498
Recently with tcpdump grab bag, found that there are a large number of drops ("packets dropped by kernel"),
As follows:
Tcpdump-i eth0 DST Port 1234 and udp-s 2048-x-tt >a.pack
Packets Captured3043 Packets received by filter2706 packets dropped by kernel
Packet Drop Reason:
After Google and analysis, the cause of this drop is due to the libcap caught
Recently, I was working on a project. Before that, I had a verification program.It was found that the client sent 1000 1024 bytes of packets consecutively, and the server experienced packet loss.The reason is that the server has not completely processed the data, and the client has sent and disabled the data.I used sleep (10) to solve this problem temporarily, but this is not a fundamental solution. If the data volume is large and the network conditio
Transferred from: http://cizixs.com/2018/01/13/linux-udp-packet-drop-debug?hmsr=toutiao.ioutm_medium=toutiao.ioutm_ Source=toutiao.ioRecent work encountered a server application UDP packet loss, in the process of reviewing a lot of information, summed up this article, for more people to refer to.Before we get started, we'll use a graph to explain the process of r
Analysis of the causes of UDP packet loss in Linux system kernel1. UDP Checksum ErrorPhenomenon: You can use NETSTAT-SU to see that there is a UDP error packet.Tcpdump catch packet, open the captured UDP message in Wireshark, turn on checksum option, error packet.Scenario: Finding link Failure www.ahlinux.com2, Firewall OpenSymptom: A package for a specific port
, we set the maximum limit of 4,096, is the driver inside each receive a batch of packets, it will ixgbe_alloc_rx_buffers assign a batch of new SKB and pagePCI_DMA mapping to the hardware to collect packetsHere the first Tasklet dispatch to CPU1, cpu2 the problem can be explained, the VM packet triggered the network card cpu1, cpu2 on the interruption, until soft interrupt, KFREE_SKB, Trigger Idx_release, then tasklet_schedule, then TX _action has bee
I don't know how to say it. In short, the boat, from the mouth, I can not see HUANGFA and impoverished! I'm not going to say anything except cursing!Prior to BBR, there are two kinds of congestion control algorithms, based on packet loss and delay-based, regardless of which is based on detection, in other words, packet loss
http://view.inews.qq.com/a/20161025A0766200QQ in narrowband eraQQ is a narrow band of the most representative of the product, in that network transmission efficiency is lower than the era, we still remember Google's homepage? Why is the simple page of Google so concise?Google was born in 1998, is also in the narrow-band era, you will find its home byte size is less than 1024, why less than 1024 bytes, because the Ethernet MTU (that is, the largest transmission unit) is 1024,google in order to le
this This paper mainly introduces the realization of WEBRTC in Nack, Weizhenwei, the article was first published in the Wind network , Id:befoioSupport original, reprint must indicate the source, welcome attention to my public number blacker (Id:blackerteam or WEBRTCORGCN).In WEBRTC, forward error correction (FEC) and packet loss retransmission (NACK) are important methods to resist network errors. FEC adds
Routing RingNetwork Packet Loss
This is an actual case of analyzing the causes of a large number of packet loss on the network. The user's network packet loss is very serious, causing a lot of trouble to the user, we try to analy
FEC (Forward error Correction) forward error correction UDP\RTP used to improve wireless and other network packet loss problems
The algorithm is not introduced at this stage.
Idea: FEC ENCODE adds redundant packets, and when the wireless network drops packets, the receiver uses redundant packets to decode the lost packets.
Example: 10 packets, the code will add 2 packets, a total of 12 packets sent to th
How should I avoid packet loss when a large amount of data is continuously transmitted over TCP?
For example, sending a file. I remember someone mentioned the possible stack overflow. How can this problem be avoided? Can I send a confirmation packet after receiving the data? After receiving the confirmation packet
The network port uses the 1000M rate time to appear the network communication loses the packet +idc the computer room managed server communication is not smooth.
Network failure:
Switch port 1000M, network card is 1000M, NIC configuration is normal. You lose the packet at the interval of ping.
The performance is packet los
Let's start by recognizing what a packet loss is, and what kind of phenomenon is being lost to the network:
Data is transmitted on the Internet on a packet-by-unit basis, with a packet of NK, no more, no less. That is to say, no matter how good the network line is, how strong the network equipment, your da
Link: http://blog.chinaunix.net/u3/105477/showart_2087878.html
Key points:Learn how to compile the sampling process, how to obtain the required parameters, how to use individual files for recording, understand the specific physical meanings of throughput, packet loss rate, and end-to-end latency, and learn more about the parameter interfaces provided by NS2! [Scenario description]: four of the eight nodes
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